GXP1610/GXP1620/GXP1625/GXP1628
Administration Guide
Page 24 of 48
Synchronization
enabled, DND, Call Forward features and Call Center Agent status can be
synchronized between Broadsoft server and phone. The default setting is
"Disabled".
Account x -> SIP Settings -> Session Timer
Enable Session Timer
To enable/disable Session Timer support.
Session Expiration
The SIP Session Timer extension that enables SIP sessions to be periodically
"refreshed" via a SIP request (UPDATE, or re-INVITE). If there is no refresh
via an UPDATE or re-INVITE message, the session will be terminated once
the session interval expires. Session Expiration is the time (in seconds) where
the session is considered timed out, provided no successful session refresh
transaction occurs beforehand. The default value is 180 seconds.
Min-SE
The minimum session expiration (in seconds). The default value is 90
seconds.
Caller Request Timer
If set to "Yes" and the remote party supports session timers, the phone will
use a session timer when it makes outbound calls.
Callee Request Timer
If set to "Yes" and the remote party supports session timers, the phone will
use a session timer when it receives inbound calls.
Force Timer
If Force Timer is set to "Yes", the phone will use the session timer even if the
remote party does not support this feature. If Force Timer is set to "No", the
phone will enable the session timer only when the remote party supports this
feature. To turn off the session timer, select "No".
UAC Specify Refresher
As a Caller, select UAC to use the phone as the refresher; or select UAS to
use the Callee or proxy server as the refresher.
UAS Specify Refresher
As a Callee, select UAC to use caller or proxy server as the refresher; or
select UAS to use the phone as the refresher.
Force INVITE
The Session Timer can be refreshed using the INVITE method or the
UPDATE method. Select "Yes" to use the INVITE method to refresh the
session timer.
Account x -> SIP Settings -> Security Settings
Check Domain
Certificates
Choose whether the domain certificates will be checked or not when TLS/TCP
is used for SIP Transport. The default setting is "No".
Validate Incoming
Messages
Choose whether the incoming messages will be validated or not. The default
setting is "No".
Check SIP User ID for
incoming INVITE
If set to "Yes", SIP User ID will be checked in the Request URI of the incoming
INVITE. If it doesn't match the phone's SIP User ID, the call will be rejected.
The default setting is "No".
Accept Incoming SIP
from Proxy Only
When set to "Yes", the SIP address of the Request URL in the incoming SIP
message will be checked. If it doesn't match the SIP server address of the
account, the call will be rejected. The default setting is "No".
Authenticate Incoming
INVITE
If set to "Yes", the phone will challenge the incoming INVITE for authentication
with SIP 401 Unauthorized response. The default setting is "No".
Account x ->Audio Settings
Send DTMF
Specifies the mechanism to transmit DTMF digits. There are 3 supported
modes: in audio which means DTMF is combined in the audio signal (not very
reliable with low-bit-rate codecs), via RTP (RFC2833), or via SIP INFO.
DTMF Payload Type
Configures the payload type for DTMF using RFC2833. The default value is
101.
Preferred Vocoder
6 different vocoder types are supported on the phone, including G.711 U-law
(PCMU), G.711 A-law (PCMA), G.723.1
(pending), G.729A/B, G.722 (wide
band), and G726-32. Users can configure vocoders in a preference list that is
